Reland "Apply global audio options at the engine level"

This is a reland of commit 1488f778a7b25c67c1c30dd9e11f9588d4b2ff38

Reland changes:
Move ownership of global audio options to ConnectionContext instead of
WebRtcVoiceEngine. That avoids an issue with a downstream implementation
where a premature Init/Terminate cycle occurred during CreateAudioSource

Original change's description:
> Apply global audio options at the engine level
>
> Introduce ApplyGlobalOptions to the MediaEngine interface to configure
> global audio processing options (such as AEC, AGC, and noise
> suppression) directly at the engine level. This replaces the previous
> behavior where media channels implicitly re-applied options to the
> engine during initialization and parameter updates.
>
> Applying global settings during CreateAudioSource ensures engine-level
> configurations persist reliably across stream lifecycles. This change
> also separates global audio processing options from stream-level
> parameters like jitter buffer and audio network adaptor settings.
>
>                  [Before Decoupling]
> +----------------------------------------------------+
> |      AudioOptions (AEC/AGC/NS + Jitter/ANA)        |
> +----------------------------------------------------+
>   /                                                \
>  / (engine()->ApplyOptions)                         \
> v                                                    v
> WebRtcVoiceEngine (Global APM)             VoiceSend/RecvChannel
>
>
>               [After Decoupling (This CL)]
> +-----------------------+                  +-----------------------+
> |  Global Options (APM) |                  |  Channel Options (ST) |
> |  - echo_cancellation  |                  |  - jitter_buffer      |
> |  - auto_gain_control  |                  |  - audio_network_     |
> |  - noise_suppression  |                  |    adaptor            |
> +-----------------------+                  +-----------------------+
>            |                                           |
>            v (ApplyGlobalOptions)                      v
>    WebRtcVoiceEngine (APM)                     VoiceSend/RecvChannel
>    - Stores in global_options_                 - No ApplyOptions calls
>    - Persisted across cycles                   - No leakage / racing
>
> Bug: webrtc:42224170
> Change-Id: I3c23293ae338495ee32330ce5287411947f6d472
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/476340
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#47844}

Bug: webrtc:42224170
Change-Id: I2e3b0374885c5f8daf25cc2fd62b56653574c13e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/476782
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#47867}
16 files changed
tree: 5426be883f728f54d62feae1980029e22a7a941f
  1. .agents/
  2. agents/
  3. api/
  4. audio/
  5. build_overrides/
  6. call/
  7. common_audio/
  8. common_video/
  9. data/
  10. docs/
  11. examples/
  12. experiments/
  13. g3doc/
  14. infra/
  15. logging/
  16. media/
  17. modules/
  18. net/
  19. p2p/
  20. pc/
  21. resources/
  22. rtc_base/
  23. rtc_tools/
  24. rust/
  25. sdk/
  26. stats/
  27. system_wrappers/
  28. test/
  29. tools_webrtc/
  30. video/
  31. .clang-format
  32. .clang-tidy
  33. .git-blame-ignore-revs
  34. .gitignore
  35. .gn
  36. .mailmap
  37. .rustfmt.toml
  38. .style.yapf
  39. .vpython3
  40. .yapfignore
  41. AUTHORS
  42. BUILD.gn
  43. CODE_OF_CONDUCT.md
  44. codereview.settings
  45. DEPS
  46. DIR_METADATA
  47. ENG_REVIEW_OWNERS
  48. GEMINI.md
  49. LICENSE
  50. license_template.txt
  51. native-api.md
  52. OWNERS
  53. OWNERS_INFRA
  54. PATENTS
  55. PRESUBMIT.py
  56. presubmit_test.py
  57. presubmit_test_mocks.py
  58. pylintrc
  59. pylintrc_old_style
  60. README.chromium
  61. README.md
  62. unsafe_buffers_paths.txt
  63. WATCHLISTS
  64. webrtc.gni
  65. webrtc_lib_link_test.cc
  66. whitespace.txt
README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See here for instructions on how to get started developing with the native code.

Authoritative list of directories that contain the native API header files.

More info